U.S. patent application number 17/130567 was filed with the patent office on 2021-04-15 for optimized coding and decoding of spatialization information for the parametric coding and decoding of a multichannel audio signal.
The applicant listed for this patent is Orange. Invention is credited to Marc Emerit, Bertrand Fatus, Stephane Ragot.
Application Number | 20210110835 17/130567 |
Document ID | / |
Family ID | 1000005300280 |
Filed Date | 2021-04-15 |
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United States Patent
Application |
20210110835 |
Kind Code |
A1 |
Fatus; Bertrand ; et
al. |
April 15, 2021 |
OPTIMIZED CODING AND DECODING OF SPATIALIZATION INFORMATION FOR THE
PARAMETRIC CODING AND DECODING OF A MULTICHANNEL AUDIO SIGNAL
Abstract
A method of parametric coding of a multichannel digital audio
signal including coding a signal arising from a channels reduction
processing applied to the multichannel signal and coding
spatialization information of the multichannel signal. The method
includes the following acts: extraction of a plurality of items of
spatialization information of the multichannel signal; obtaining at
least one representation model of the extracted spatialization
information; determination of at least one angle parameter of a
model obtained; coding the at least one determined angle parameter
so as to code the spatialization information extracted during the
coding of spatialization information. Also provided are a method
for decoding such a coded signal and corresponding coding and
decoding devices.
Inventors: |
Fatus; Bertrand; (Le
Chesnay, FR) ; Ragot; Stephane; (Lannion, FR)
; Emerit; Marc; (Rennes, FR) |
|
Applicant: |
Name |
City |
State |
Country |
Type |
Orange |
Paris |
|
FR |
|
|
Family ID: |
1000005300280 |
Appl. No.: |
17/130567 |
Filed: |
December 22, 2020 |
Related U.S. Patent Documents
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Application
Number |
Filing Date |
Patent Number |
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16083741 |
Sep 10, 2018 |
10930290 |
|
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PCT/FR2017/050547 |
Mar 10, 2017 |
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17130567 |
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Current U.S.
Class: |
1/1 |
Current CPC
Class: |
G10L 25/18 20130101;
G10L 19/008 20130101 |
International
Class: |
G10L 19/008 20060101
G10L019/008; G10L 25/18 20060101 G10L025/18 |
Foreign Application Data
Date |
Code |
Application Number |
Mar 10, 2016 |
FR |
1652034 |
Claims
1. A method comprising: parametric decoding a multichannel digital
audio signal comprising the following acts performed by a decoding
device: decoding a signal arising from a channels reduction
processing applied to the multichannel and coded signal; and
decoding spatialization cues in respect of the multichannel signal,
comprising: receiving at least one coded angle parameter from a
communication network or reading the at least one coded angle
parameter from a non-transitory computer-readable medium; decoding
the at least one coded angle parameter; obtaining at least one
representation model of spatialization cues; and determining a
plurality of spatialization cues in respect of the multichannel
signal on the basis of the at least one model obtained and of the
at least one decoded angle parameter.
2. The method as claimed in claim 1, wherein the spatialization
cues are defined by frequency sub-bands of the multichannel audio
signal and at least one angle parameter per sub-band is
received.
3. The method as claimed in claim 1, wherein the method furthermore
comprises receiving a reference spatialization cue and decoding
this reference spatialization cue.
4. The method as claimed in claim 1, wherein one of the
spatialization cues is an interchannel time shift (ITD) cue.
5. The method as claimed in claim 1, wherein one of the
spatialization cues is an interchannel intensity difference (ILD)
cue.
6. The method as claimed in claim 5, wherein the method furthermore
comprises the following acts for decoding an interchannel intensity
difference cue: estimating an interchannel intensity difference cue
on the basis of the model obtained and of the angle parameter;
decoding the difference between the interchannel intensity
difference cue.
7. The method as claimed in claim 1, wherein a
spatialization-cue-based representation model is obtained.
8. The method as claimed in claim 1, wherein a representation model
common to several spatialization cues is obtained.
9. The method as claimed in claim 1, further comprising receiving
and decoding an index of a table of models and obtaining the at
least one representation model of the spatialization cues to be
decoded on the basis of the decoded index.
10. A parametric decoder of a multichannel digital audio signal,
comprising: a processor; and a non-transitory computer-readable
medium comprising instructions stored thereon, which when executed
by the processor configure the parametric decoder to perform acts
to parametric decode the multichannel digital audio signal;
decoding a signal arising from a channels reduction processing
applied to the multichannel and coded signal; and decoding
spatialization cues in respect of the multichannel signal,
comprising: receiving at least one coded angle parameter from a
communication network or reading the at least one coded angle
parameter from a storage medium; decoding the at least one coded
angle parameter; obtaining at least one representation model of
spatialization cues; and determining a plurality of spatialization
cues in respect of the multichannel signal on the basis of the at
least one model obtained and of the at least one decoded angle
parameter.
11. A non-transitory computer-readable medium on which is recorded
a computer program comprising code instructions for execution of a
method of parametric decoding a multichannel digital audio signal
when the instructions are executed by a processor of a decoding
device, wherein the method comprises: decoding a signal arising
from a channels reduction processing applied to the multichannel
and coded signal; and decoding spatialization cues in respect of
the multichannel signal, comprising: receiving at least one coded
angle parameter from a communication network or reading the at
least one coded angle parameter from a storage medium; decoding the
at least one coded angle parameter; obtaining at least one
representation model of spatialization cues; and determining a
plurality of spatialization cues in respect of the multichannel
signal on the basis of the at least one model obtained and of the
at least one decoded angle parameter.
Description
CROSS-REFERENCE TO RELATED APPLICATIONS
[0001] This application is divisional of U.S. application Ser. No.
16/083,741, filed Sep. 10, 2018, which is a Section 371 National
Stage Application of International Application No.
PCT/FR2017/050547, filed Mar. 10, 2017, and published as WO
2017/153697 on Sep. 14, 2017, not in English, the contents of which
are incorporated herein by reference in their entireties.
FIELD OF THE DISCLOSURE
[0002] The present invention relates to the field of the
coding/decoding of digital signals.
[0003] The coding and the decoding according to the invention is
adapted in particular to the transmission and/or the storage of
digital signals such as audiofrequency signals (speech, music or
other).
[0004] More particularly, the present invention pertains to the
parametric multichannel coding and decoding of multichannel audio
signals.
[0005] The invention is therefore concerned with multichannel
signals, and in particular with binaural signals which are sound
signals recorded with microphones placed at the entrance of the
canal of each ear (of a person or of a mannequin) or else
synthesized artificially by way of filters known as HRIR
(Head-Related Impulse Response) filters in the time domain or HRTF
(Head-Related Transfer Function) filters in the frequency domain,
which are dependent on the direction and distance of the sound
source and the morphology of the subject.
BACKGROUND OF THE DISCLOSURE
[0006] Binaural signals are associated with listening typically
with a headset or earpiece and exhibit the advantage of
representing a spatial image giving the illusion of being naturally
in the midst of a sound scene; it therefore entails reproduction of
the sound scene in 3D with only 2 channels. It will be noted that
it is possible to listen to a binaural sound on loudspeakers by way
of complex processings for inverting the HRIR/HRTF filters and for
reconstructing binaural signals.
[0007] Here we distinguish binaural signals from stereo signals. A
stereo signal also consists of two channels but it does not in
general allow perfect reproduction of the sound scene in 3D. For
example, a stereo signal can be constructed by taking a given
signal on the left channel and a zero signal on the right channel,
listening to such a signal will give a sound source location on the
left but in a natural environment this stratagem is not possible
since the signal to the right ear is a filtered version (including
a time shift and an attenuation) of the signal to the left ear as a
function of the person's morphology.
[0008] Parametric multichannel coding is based on the extraction
and the coding of spatial-information parameters so that, on
decoding, these spatial characteristics can be used to recreate the
same spatial image as in the original signal. Examples of codecs
based on this principle are found in the 3GPP e-AAC+ or MPEG
Surround standards.
[0009] The case of parametric stereo coding with N=2 channels is
considered here by way of example, insofar as its description is
simpler than in the case of N>2 channels.
[0010] A parametric stereo coding/decoding technique is for example
described in the document by J. Breebaart, S. van de Par, A.
Kohlrausch, E. Schuijers, entitled "Parametric Coding of Stereo
Audio" in EURASIP Journal on Applied Signal Processing 2005:9, pp.
1305-1322. This example is also employed with reference to FIGS. 1
and 2 describing respectively a parametric stereo coder and
decoder.
[0011] Thus, FIG. 1 describes a stereo coder receiving two audio
channels, a left channel (denoted L for Left in English) and a
right channel (denoted R for Right in English).
[0012] The temporal signals L (n) and R (n), where n is the integer
index of the samples, are processed by the blocks 101, 102, 103 and
104 which perform a short-term Fourier analysis. The transformed
signals L[k] and R[k], where k is the integer index of the
frequency coefficients, are thus obtained.
[0013] The block 105 performs a channels reduction processing or
"downmix" in English to obtain in the frequency domain on the basis
of the left and right signals, a monophonic signal hereinafter
named mono signal. Several techniques have been developed for
stereo to mono channel reduction or "downmix" processing. This
"downmix" can be performed in the time or frequency domain. One
generally distinguishes: [0014] Passive "downmix" which corresponds
to a direct matrixing of the stereo channels to combine them into a
single signal--the coefficients of the downmix matrix are in
general real and of predetermined (fixed) values; [0015] Active
(adaptive) "downmix" which includes control of the energy and/or of
the phase in addition to the combining of the two stereo
channels.
[0016] Extraction of spatial-information parameters is also
performed in the block 105. The extracted parameters are the
following.
[0017] The parameters ICLD or ILD or CLD (for "InterChannel/Channel
Level Difference" in English), also called differences of
interchannel intensity, characterize the ratios of energy per
frequency sub-band between the left and right channels. These
parameters make it possible to position sound sources in the stereo
horizontal plane by "panning". They are defined in dB by the
following formula:
ICLD [ b ] = 10 log 10 { .SIGMA. k = k b k b + 1 - 1 L [ k ] L * [
k ] .SIGMA. k = k b k b + 1 - 1 R [ k ] R * [ k ] } dB ( 1 )
##EQU00001##
where L[k] and R[k] correspond to the (complex) spectral
coefficients of the channels L and R, each frequency band of index
b=0, . . . , B-1 comprises the frequency spectral lines in the
interval [k.sub.b, k.sub.b+1-1], the symbol * indicates the complex
conjugate and B is the number of sub-bands.
[0018] The parameters ICPD or IPD (for "InterChannel Phase
Difference" in English), also called phase differences, are defined
according to the following relation:
ICPD[b]=.angle.(.SIGMA..sub.k=k.sub.b.sup.k.sup.b+1.sup.-1L[k],R*[k])
(2)
where .angle. indicates the argument (the phase) of the complex
operand. It is also possible to define in an equivalent manner to
the ICPD, an interchannel time shift called ICTD or ITD (for
"InterChannel Time Difference" in English). The ITD can for example
be measured as the delay which maximizes the intercorrelation
between L and R:
ITD = max - d .ltoreq. .tau. .ltoreq. d n = 0 N - .tau. - 1 L ( n +
.tau. ) R ( n ) ( 3 ) ##EQU00002##
where d defines the search interval for the maximum. It will be
noted that the correlation in equation (3) can be normalized.
[0019] In contradistinction to the parameters ICLD, ICPD and ICTD
which are location parameters, the parameter ICC (for "InterChannel
Coherence" in English) represents the level of inter-channel
correlation (or coherence) and is associated with the spatial width
of a sound source; the ICC can be defined as:
ICC = max - d .ltoreq. .tau. .ltoreq. d n = 0 N - .tau. - 1 L ( n +
.tau. ) R ( n ) ( 4 ) ##EQU00003##
[0020] where the correlation can be normalized just as for eq.
(3).
[0021] It is noted in the article by Breebart et al. that the ICC
parameters are not necessary in the sub-bands that are reduced to a
single frequency coefficient--indeed the differences of amplitude
and of phase completely describe the spatialization in this
"degenerate" case.
[0022] The ICLD and ICPD parameters are extracted by analysis of
the stereo signals, by the block 105. The ICTD or ICC parameters
can also be extracted per sub-band on the basis of the spectra L[k]
and R[k]; however their extraction is in general simplified by
assuming an identical interchannel time shift for each sub-band and
in this case a parameter can be extracted on the basis of the
temporal channels L(n) and R(n).
[0023] The mono signal M[k] is transformed into the time domain
(blocks 106 to 108) after short-term Fourier synthesis (inverse
FFT, windowing and OverLap-Add or OLA in English) and a mono coding
(block 109) is carried out thereafter. In parallel the stereo
parameters are quantized and coded in the block 110.
[0024] In general the spectrum of the signals (L[k], R[k]) is
divided according to a non-linear frequency scale of ERB
(Equivalent Rectangular Bandwidth) or Bark type. The parameters
(ICLD, ICPD, ICC, ITD) are coded by scalar quantization optionally
followed by an entropy coding and/or by a differential coding. For
example, in the article cited above, the ICLD is coded by a
non-uniform quantizer (ranging from -50 to +50 dB) with
differential entropy coding. The non-uniform quantization step
exploits the fact that the larger the value of the ICLD the lower
the auditory sensitivity to the variations of this parameter.
[0025] For the coding of the mono signal (block 109), several
quantization techniques with or without memory are possible, for
example "Pulse-Code Modulation" (PCM) coding, its version with
adaptive prediction termed "Adaptive Differential Pulse-Code
Modulation" (ADPCM) or more advanced techniques such as
transform-based perceptual coding or "Code Excited Linear
Prediction" (CELP) coding or multi-mode coding.
[0026] One is concerned here more particularly with the 3GPP EVS
(for "Enhanced Voice Services") standard which uses multi-mode
coding. The algorithmic details of the EVS codec are provided in
the specifications 3GPP TS 26.441 to 26.451 and they are therefore
not repeated here. Hereinafter, these specifications will be
referred to by the name EVS.
[0027] The input signal of the (mono) EVS codec is sampled at the
frequency of 8, 16, 32 or 48 kHz and the codec can represent
telephone audio bands (narrowband, NB), wide (wideband, WB),
super-wide (super-wideband, SWB) or full band (fullband, FB). The
bitrates of the EVS codec are divided into two modes: [0028] "EVS
Primary": [0029] fixed bitrates: 7.2, 8, 9.6, 13.2, 16.4, 24.4, 32,
48, 64, 96, 128 [0030] variable bitrate (VBR) mode with a mean
bitrate close to 5.9 kbit/s for active speech [0031]
"channel-aware" mode at 13.2 in WB and SWB only [0032] "EVS AMR-WB
TO" whose bitrates are identical to the AMR-WB 3GPP codec (9
modes)
[0033] To this is added the discontinuous-transmission mode (DTX)
in which the frames detected as inactive are replaced with SID
frames (SID Primary or SID AMR-WB TO) which are transmitted in an
intermittent manner, about once every 8 frames.
[0034] At the decoder 200, with reference to FIG. 2, the mono
signal is decoded (block 201), a decorrelator is used (block 202)
to produce two versions {circumflex over (M)}(n) and {circumflex
over (M)}'(n) of the decoded mono signal. This decorrelation,
necessary only when the parameter ICC is used, makes it possible to
increase the spatial width of the mono source {circumflex over
(M)}(n). These two signals {circumflex over (M)}(n) and {circumflex
over (M)}'(n) are passed into the frequency domain (blocks 203 to
206) and the decoded stereo parameters (block 207) are used by the
stereo synthesis (or shaping) (block 208) to reconstruct the left
and right channels in the frequency domain. These channels are
finally reconstructed in the time domain (blocks 209 to 214).
[0035] An exemplary parametric stereo coding seeking to represent
binaural signals (without regard for the nature of the HRTF
filters) is described in the article by Pasi Ojala, Mikko Tammi,
Miikka Vilermo, entitled "Parametric binaural audio coding", in
Proc. ICASSP, 2010, pp. 393-396. Two parameters are coded to
restore a spatial image with a location close to a binaural image:
the ICLD and the ITD. Moreover a parameter ALC (for "Ambience Level
Control" in English) similar to the ICC is also coded, making it
possible to control the level of the "ambience" associated with the
use of decorrelated channels. This codec is described for signals
in the super-wide band with 20-ms frames and a bitrate of 20 or 32
kbit/s to code the mono signal to which is added a bitrate of 5
kbit/s to code the spatial parameters.
[0036] Another exemplary parametric stereo codec developed with a
specific mode to code binaural signals is given by the standard
G.722 Annex D, in particular in the stereo coding mode R1ws in the
widened band to 56+8 kbit/s. This codec operates with "short"
frames of 5 ms according to 2 modes: a "transient" mode where ICLDs
are coded on 38 bits and a "normal" mode where ICLDs are coded on
24 bits with a full-band ITD/IPD on 5 bits. The details of
estimating the ITD, of coding the ICLD and ITD parameters are not
repeated here. It will be noted that the ICLDs are coded by
"decimation" by distributing the coding of the ICLDs over several
successive frames, coding only a subset of the parameters of a
given frame.
[0037] In the two examples it is important to note that one is not
dealing with binaural codecs, but with stereo codecs seeking to
reproduce a spatial image similar to a binaural signal.
[0038] It will be noted that the case of parametric multichannel
coding with N>2 follows the same principle as the case N=2,
however in general the downmix might not be mono but stereo and the
inter-channel parameters must cover more than 2 channels. An
exemplary embodiment is given in the MPEG Surround standard where
ICLD, ICTD and ICC parameters are coded. It will also be noted that
the MPEG Surround decoder includes a binaural restoration,
parametrized by HRTF filters.
[0039] Let us consider now the case of a stereo coding and decoding
of parameters of ICLD type such as is described in FIGS. 1 and 2
and let us take the case of a signal in the widened band, sampled
at 16 kHz and analyzed with frames of 20 ms and a sinusoidal
windowing covering 40 ms (including 20 ms of "lookahead"). For the
extraction of the ICLD parameters (block 105), the spectra L[k] and
R[k] may be for example sliced into B frequency sub-bands according
to the ERB scale. For each frame, the ICLD of the sub-band b=0, . .
. , 34 is calculated according to the equation:
ICLD [ b ] = 10 log 1 0 { .sigma. L 2 [ b ] .sigma. R 2 [ b ] } ( 5
) ##EQU00004##
where .sigma..sub.L.sup.2[b] and .sigma..sub.R.sup.2[b] represent
respectively the energy of the left channel (L[k]) and of the right
channel (R[k]):
{ .sigma. L 2 [ b ] = k = k b k b + 1 - 1 L [ k ] L * [ k ] .sigma.
R 2 [ b ] = k = k b k b + 1 - 1 R [ k ] R * [ k ] ( 6 )
##EQU00005##
[0040] According to the prior art, the coding of a block of 35 ICLD
of a given frame can be carried out for example with: [0041] 5 bits
for the first ICLD parameter (coded in absolute), [0042] 4 bits for
the following 32 ICLD parameters (coded in differential), [0043] 3
bits for the last 2 ICLD parameters (coded in differential).
[0044] thus giving a total of 5+32.times.4+2.times.3=139
bits/frame, i.e. a bitrate of close to 7 kbit/s in the case of
20-ms frames. This bitrate does not comprise the other
parameters.
[0045] This bitrate of approximately 7 kbit/s can be reduced on
average by using a variable-bitrate entropy coding, for example a
Huffman coding; however, in most cases, a drastic bitrate reduction
will not be possible.
[0046] To halve the bitrate of the coding of the ICLD parameters,
it would be possible to use the alternate coding approach described
previously in the case of stereo G.722 coding. However, the
associated bitrate remains significant for a coding with 35
sub-bands and 20 ms of frame; moreover, the temporal resolution of
the coding would be reduced and this may be problematic in the case
of non-stationary signals. Another approach would consist in
reducing the number of sub-bands to go from 35 to for example 20
sub-bands. This would reduce the bitrate associated with the ICLD
parameters, but would in general degrade the fidelity of the
synthesized spatial image.
[0047] If it is assumed that the coder of FIG. 1 is a stereo coder
operating for example at bitrates of 16.4, 24.4, 32, 48, 64, 96,
128 kbit/s and that it relies on a downmix coded by a mono EVS
codec, then for the lowest bitrates, for example 16.4 kbit/s in
stereo, if the downmix is coded with the mono EVS codec at 13.2
kbit/s, then only 3.2 kbit/s remain to code all the spatial
parameters in order to faithfully represent a spatial image. If it
is necessary to code not only ICLD parameters, but also other
spatial parameters, it is understood that the previously described
coding of the ICLD parameters requires too much bitrate.
[0048] A need therefore exists to represent the spatial parameters
of a multichannel signal in an efficient manner, at as low a
bitrate as possible and with acceptable quality.
SUMMARY
[0049] The invention improves the situation of the prior art.
[0050] For this purpose, it proposes a method of parametric coding
of a multichannel digital audio signal comprising a step of coding
a signal arising from a channels reduction processing applied to
the multichannel signal and of coding spatialization cues in
respect of the multichannel signal. The method is such that it
comprises the following steps: [0051] extraction of a plurality of
spatialization cues in respect of the multichannel signal; [0052]
obtaining of at least one representation model of the
spatialization cues extracted; [0053] determination of at least one
angle parameter of a model obtained; [0054] coding of the at least
one determined angle parameter so as to code the spatialization
cues extracted during the coding of spatialization cues.
[0055] The scheme for coding the spatialization cues relies on a
model-based approach which makes it possible to approximate the
spatial cues. Thus the coding of a plurality of spatial cues is
reduced to the coding of an angle parameter thereby considerably
reducing the coding bitrate with respect to the direct coding of
the spatial cue. The bitrate required for the coding of this
parameter is therefore reduced.
[0056] In a particular embodiment based on sub-bands, the
spatialization cues are defined by frequency sub-bands of the
multichannel audio signal and at least one angle parameter per
sub-band is determined and coded.
[0057] In a particular embodiment, the method furthermore comprises
the steps of calculating a reference spatialization cue and of
coding this reference spatialization cue.
[0058] Thus, the coding of a reference cue can improve decoding
quality. The bitrate for coding this reference cue does not require
too significant a bitrate.
[0059] This scheme is particularly well suited to the coding of the
spatial cue of interchannel time shift (ITD) type and/or of
interchannel intensity difference (ILD) type.
[0060] To further improve the quality of decoding of the cue of ILD
type, the method furthermore comprises the following steps: [0061]
estimation of an interchannel intensity difference cue on the basis
of the model obtained and of the angle parameter determined; [0062]
coding of the difference between the interchannel intensity
difference cue extracted and estimated.
[0063] The coding of this residual requires an additional coding
bitrate but this scheme still affords a gain in bitrate with
respect to the direct coding of the ILD spatialization cue.
[0064] In a particular embodiment, a spatialization-cue-based
representation model is obtained. It can be fixed and stored in
memory.
[0065] This fixed and recorded model is for example a model of sine
form. This type of model is adapted to suit the form of the ITD or
ILD cue according to the position of the source.
[0066] In a variant embodiment, the obtaining of a representation
model of the spatialization cues is performed by selecting from a
table of models defined for various values of the spatialization
cues.
[0067] Several models may be selectable as a function of
characteristics of the multichannel signal. This makes it possible
to best adapt the spatialization cue model to the signal.
[0068] The index of the model chosen can then be in one embodiment,
coded and transmitted.
[0069] In a variant embodiment a representation model common to
several spatialization cues is obtained.
[0070] This makes it possible to pool the selection of a model to
several spatialization cues, thereby reducing the processing
operations to be performed.
[0071] The invention also pertains to a method of parametric
decoding of a multichannel digital audio signal comprising a step
of decoding a signal arising from a channels reduction processing
applied to the multichannel and coded signal and of decoding
spatialization cues in respect of the multichannel signal. The
method is such that it comprises the following steps for decoding
at least one spatialization cue: [0072] reception and decoding of
at least one coded angle parameter; [0073] obtaining of at least
one representation model of spatialization cues; [0074]
determination of a plurality of spatialization cues in respect of
the multichannel signal on the basis of the at least one model
obtained and of the at least one decoded angle parameter.
[0075] In the same way as for the coding, this scheme based on the
use of a representation model of the spatialization cues makes it
possible to retrieve the cue with good quality without it being
necessary to have too large a bitrate. At reduced bitrate, a
plurality of spatialization cues is retrieved by decoding a simple
angle parameter.
[0076] In a particular embodiment, the method comprises a step of
receiving and decoding an index of table of models and of obtaining
the at least one representation model of the spatialization cues to
be decoded on the basis of the decoded index.
[0077] Thus, it is possible to adapt the model to be used according
to the characteristics of the multichannel signal.
[0078] The invention pertains to a parametric coder of a
multichannel digital audio signal comprising a module for coding a
signal arising from a module for channels reduction processing
applied to the multichannel signal and modules for coding
spatialization cues in respect of the multichannel signal. The
coder is such that it comprises: [0079] a module for extracting a
plurality of spatialization cues in respect of the multichannel
signal; [0080] a module for obtaining at least one representation
model of the spatialization cues extracted; [0081] a module for
determining at least one angle parameter of a model obtained;
[0082] a module for coding the at least one angle parameter
determined so as to code the spatialization cues extracted during
the coding of spatialization cues.
[0083] The coder exhibits the same advantages as the method that it
implements.
[0084] The invention pertains to a parametric decoder of a
multichannel digital audio signal comprising a module for decoding
a signal arising from a channels reduction processing applied to
the multichannel and coded signal and a module for decoding
spatialization cues in respect of the multichannel signal. The
decoder is such that it comprises: [0085] a module for receiving
and decoding at least one coded angle parameter; [0086] a module
for obtaining at least one representation model of the
spatialization cues; [0087] a module for determining a plurality of
spatialization cues in respect of the multichannel signal on the
basis of the at least one model obtained and of the at least one
decoded angle parameter.
[0088] The decoder exhibits the same advantages as the method that
it implements.
[0089] Finally, the invention pertains to a computer program
comprising code instructions for the implementation of the steps of
a coding method according to the invention, when these instructions
are executed by a processor, to a computer program comprising code
instructions for the implementation of the steps of a decoding
method according to the invention, when these instructions are
executed by a processor.
[0090] The invention pertains finally to storage medium readable by
a processor on which is recorded a computer program comprising code
instructions for the execution of the steps of the coding method
such as described and/or of the decoding method such as
described.
BRIEF DESCRIPTION OF THE DRAWINGS
[0091] Other characteristics and advantages of the invention will
become more clearly apparent on reading the following description,
given solely by way of nonlimiting example and with reference to
the appended drawings in which:
[0092] FIG. 1 illustrates a coder implementing a parametric coding
known from the prior art and described previously;
[0093] FIG. 2 illustrates a decoder implementing a parametric
decoding known from the prior art and described previously;
[0094] FIG. 3 illustrates a parametric coder according to one
embodiment of the invention;
[0095] FIGS. 4a, 4b and 4c illustrate the steps of the coding
method according to various embodiments of the invention by a
detailed illustration of the blocks for coding spatial cues;
[0096] FIGS. 5a, 5b illustrate the notions of sound perception in
3D and 2D and FIG. 5c illustrates a schematic representation of
polar coordinates (distance, azimuth) of an audio source in the
horizontal plane with respect to a listener, in the binaural
case;
[0097] FIG. 6a illustrates representations of models of total
energy of HRTFs suitable for representing spatial cues of ILD
type;
[0098] FIG. 6b illustrates a configuration of stereo microphones of
ORTF type picking up an exemplary signal with two channels to be
coded according to one embodiment of the coding method of the
invention;
[0099] FIGS. 6c to 6g illustrate representations of a cue model
M.sub.ILD (m, t) (for m=0 and t corresponding to an azimuth from 0
to 360.degree.) of spatialization of ILD type by sub-bands in a 1/3
octave slicing, as a function of the azimuth angle;
[0100] FIG. 7 illustrates a parametric decoder as well as the
decoding method according to one embodiment of the invention;
[0101] FIG. 8 illustrates a variant embodiment of a parametric
coder according to the invention;
[0102] FIG. 9 illustrates a variant embodiment of a parametric
decoder according to the invention; and
[0103] FIG. 10 illustrates a hardware example of an item of
equipment incorporating a coder able to implement the coding method
according to one embodiment of the invention or a decoder able to
implement the decoding method according to one embodiment of the
invention.
DETAILED DESCRIPTION OF ILLUSTRATIVE EMBODIMENTS
[0104] With reference to FIG. 3, a parametric coder of signals with
two channels according to one embodiment of the invention,
delivering both a mono binary train and spatial-information
parameters in respect of the input signal is now described. This
figure presents at one and the same time the entities, hardware or
software modules driven by a processor of the coding device and the
steps implemented by the coding method according to one embodiment
of the invention.
[0105] The case of a signal with two channels is described here.
The invention also applies to the case of a multichannel signal
with a number of channels greater than 2.
[0106] To avoid overburdening the text, the coder described in FIG.
3 will be called a "stereo coder" even if it allows the coding of
binaural signals. Likewise the parameters ICLD, ICTD, ICPD will be
respectively denoted ILD, ITD, IPD even if the signal is not
binaural.
[0107] This parametric stereo coder such as illustrated uses an EVS
mono coding according to the specifications 3GPP TS 26.442
(fixed-point source code) or TS 26.443 (floating-point source
code), it operates with stereo or multichannel signals sampled at
the sampling frequency F.sub.s of 8, 16, 32 and 48 kHz, with 20-ms
frames. Hereinafter, with no loss of generality, the description is
given mainly for the case F.sub.s=16 kHz and for the case N=2
channels.
[0108] It should be noted that the choice of a frame length of 20
ms is not in any case restrictive in the invention which applies
likewise in variants of the embodiment where the frame length is
different, for example 5 or 10 ms, with a codec other than EVS.
[0109] Moreover, the invention applies likewise to other types of
mono coding (e.g.: IETF OPUS, UIT-T G.722) operating at identical
or non-identical sampling frequencies.
[0110] Each temporal channel (L(n) and R(n)) sampled at 16 kHz is
firstly pre-filtered by a high-pass filter (HPF for High Pass
Filter in English) typically eliminating the components below 50 Hz
(blocks 301 and 302). This pre-filtering is optional, but it can be
used to avoid the bias due to the continuous component (DC) in the
estimation of parameters such as the ICTD or the ICC.
[0111] The channels L'(n) and R'(n) arising from the pre-filtering
blocks are analyzed in terms of frequencies by discrete Fourier
transform with sinusoidal windowing with overlap of 50% of length
40 ms i.e. 640 samples (blocks 303 to 306). For each frame, the
signal (L'(n), R'(n)) is therefore weighted by a symmetric analysis
window covering 2 frames of 20 ms i.e. 40 ms (or 640 samples for
F.sub.s=16 kHz). The 40-ms analysis window covers the current frame
and the future frame. The future frame corresponds to a "future"
signal segment commonly called "lookahead" of 20 ms. In variants of
the invention, other windows could be used, for example a low-delay
asymmetric window called "ALDO" in the EVS codec. Moreover, in
variants, the analysis windowing could be rendered adaptive as a
function of the current frame, so as to use an analysis with a long
window on stationary segments and an analysis with short windows on
transient/non-stationary segments, optionally with transition
windows between long and short windows.
[0112] For the current frame of 320 samples (20 ms at F.sub.s=16
kHz), the spectra obtained, L[k] and R[k] (k=0 . . . 320), comprise
321 complex coefficients, with a resolution of 25 Hz per frequency
coefficient. The coefficient of index k=0 corresponds to the
continuous component (0 Hz), it is real. The coefficient of index
k=320 corresponds to the Nyquist frequency (8000 Hz for F.sub.s=16
kHz), it is also real. The coefficients of index 0<k<160 are
complex and correspond to a sub-band of width 25 Hz centered on the
frequency of k.
[0113] The spectra L[k] and R[k] are combined in the block 307 to
obtain a mono signal (downmix) M[k] in the frequency domain. This
signal is converted into time by inverse FFT and windowing-overlap
with the "lookahead" part of the previous frame (blocks 308 to
310).
[0114] An example of frequency "downmix" technique is described in
the document entitled "A stereo to mono downmixing scheme for
MPEG-4 parametric stereo encoder" by Samsudin, E. Kurniawati, N.
Boon Poh, F. Sattar, S. George, in Proc. ICASSP, 2006.
[0115] In this document, the L and R channels are aligned in phase
before performing the channels reduction processing.
[0116] More precisely, the phase of the L channel for each
frequency sub-band is chosen as the reference phase, the R channel
is aligned according to the phase of the L channel for each
sub-band through the following formula:
R'[k]=e.sup.jICPD[b]R[k] (7)
where R'[k] is the aligned R channel, k is the index of a
coefficient in the b.sup.th frequency sub-band, ICPD[b] is the
inter-channel phase difference in the b.sup.th frequency sub-band
given by equation (2).
[0117] Note that when the sub-band of index b is reduced to a
frequency coefficient, we find:
R'[k]=|R[k]|e.sup.j.angle.L[k] (8)
[0118] Finally the mono signal obtained by the "downmix" of the
document of Samsudin et al. cited previously is calculated by
averaging the L channel and the aligned R' channel, according to
the following equation:
M [ k ] = L [ k ] + R ' [ k ] 2 ( 9 ) ##EQU00006##
[0119] The phase alignment therefore makes it possible to preserve
the energy and to avoid the problems of attenuation by eliminating
the influence of the phase. This "downmix" corresponds to the
"downmix" described in the document by Breebart et al. where:
M[k]=w.sub.1L[k]+w.sub.2R[k] (10)
with w.sub.1=0.5 and
w 2 = e j . ICPD [ b ] 2 ##EQU00007##
in the case where the sub-band of index b comprises only a
frequency value of index k.
[0120] Other "downmix" schemes can of course be chosen without
modifying the scope of the invention.
[0121] The algorithmic delay of the EVS codec is 30.9375 ms at
F.sub.s=8 kHz and 32 ms for the other frequencies F.sub.s=16, 32 or
48 kHz. This delay includes the current frame of 20 ms, the
additional delay with respect to the frame length is therefore
10.9375 ms at F.sub.s=8 kHz and 12 ms for the other frequencies
(i.e. 192 samples at F.sub.s=16 kHz), the mono signal is delayed
(block 311) by T=320-192=128 samples so that the delay accumulated
between the mono signal decoded by EVS and the original stereo
channels becomes a multiple of the length of frames (320 samples).
Accordingly, to synchronize the extraction of stereo parameters
(block 314) and the spatial synthesis on the basis of the mono
signal performed at the decoder, the lookahead for the calculation
of the mono signal (20 ms) and the mono coding/decoding delay to
which is added the delay T to align the mono synthesis (20 ms)
correspond to an additional delay of 2 frames (40 ms) with respect
to the current frame. This delay of 2 frames is specific to the
implementation detailed here, in particular it is related to the
20-ms sinusoidal symmetric windows. This delay could be different.
In a variant embodiment, it would be possible to obtain a delay of
a frame with an optimized window with a smaller overlap between
adjacent windows with a block 311 not introducing any delay
(T=0).
[0122] The shifted mono signal is thereafter coded (block 312) by
the mono EVS coder for example at a bitrate of 13.2, 16.4 or 24.4
kbit/s. In variants, the coding could be performed directly on the
unshifted signal; in this case the shift could be performed after
decoding.
[0123] In a particular embodiment of the invention, illustrated
here in FIG. 3, it is considered that the block 313 introduces a
delay of two frames on the spectra L[k], R[k] and M[k] so as to
obtain the spectra L.sub.buf[k], R.sub.buf[k] and M.sub.buf[k].
[0124] It would be possible in a more advantageous manner in terms
of quantity of data to be stored, to shift the outputs of the
parameters extraction block 314 or else the outputs of the
quantization blocks 318, 316 and 319. It would also be possible to
introduce this shift at the decoder on receiving the binary train
of the stereo coder.
[0125] In parallel with the mono coding, the coding of the spatial
cue is implemented in the blocks 315 to 319 according to a coding
method of the invention. Moreover, the coding comprises an optional
step of classifying the input signal in the block 321.
[0126] This classification block, according to the multichannel
signal to be coded, can make it possible to pass from one mode of
coding to another. One of the coding modes being that implementing
the invention for the coding of the spatialization cues. The other
coding modes are not detailed here, but it will be possible to use
conventional techniques for stereo or multichannel coding,
including techniques for parametric coding with ILD, ITD, IPD, ICC
parameters. The classification is indicated here with the L and R
temporal signals as input, optionally the signals in the frequency
domain and the stereo or multichannel parameters will also be able
to serve for the classification. It will also be possible to use
the classification to apply the invention to a given spatial
parameter (for example to code the ITD or the ILD), stated
otherwise to switch the type of coding of spatial parameters with a
possible choice between a coding scheme according to a model as in
the invention or an alternative coding scheme of the prior art.
[0127] The spatial parameters are extracted (block 314) on the
basis of the spectra L[k], R[k] and M[k] shifted by two frames:
L.sub.buf[k], R.sub.buf[k] and M.sub.buf[k] and coded (blocks 315
to 319) according to a coding method described with reference to
FIGS. 4a to 4c and detailing the blocks 315 and 317.
[0128] For the extraction of the parameters ILD (block 314), the
spectra L.sub.buf[k] and R.sub.buf[k] are for example sliced into
frequency sub-bands.
[0129] In one embodiment, a 1/3 octave sub-band slicing defined in
array 1 hereinbelow will be taken:
TABLE-US-00001 No Octave Thirds 1 2 3 4 5 6 7 8 9 10 11 12 Base
frequency (Hz) 0 111 140 177 223 281 354 445 561 707 891 1122 High
Frequency (Hz) 111 140 177 223 281 354 445 561 707 891 1122 1414 No
Octave Thirds 13 14 15 16 17 18 19 20 21 22 23 24 Base frequency
(Hz) 1414 1782 2245 2828 3564 4490 5657 7127 8980 11314 14254 17959
High Frequency (Hz) 1782 2245 2828 3564 4490 5657 7127 8980 11314
14254 17959 22627
[0130] Array 1
[0131] This array covers all the cases of sampling frequency, for
example for a coder with a sampling frequency at 16 kHz only the
first B=20 sub-bands will be retained. Thus, it will be possible to
define the array:
TABLE-US-00002 k.sub.b = 0 . . . 20 = [0 4 6 7 9 11 14 18 22 28 36
45 57 71 90 113 143 180 226 285 320]
[0132] The above array delimits (as index of Fourier spectral
lines) the frequency sub-bands of index b=0 to B-1 for the case
F.sub.s=16 kHz. Each sub-band of index b comprises the coefficients
k.sub.b=0 to k.sub.b+1-1. The frequency spectral line of index
k=320 which corresponds to the Nyquist frequency is not taken into
account here.
[0133] In variants, it will be possible to use another sub-band
slicing, for example according to the ERB scale; in this case, it
will be possible to use B=35 sub-bands, the latter are defined by
the following boundaries in the case where the input signal is
sampled at 16 kHz:
TABLE-US-00003 k.sub.b = 0 . . . 35 = [0 1 2 3 5 6 8 10 12 14 17 20
23 27 31 35 40 46 52 58 66 74 83 93 104 117 130 145 162 181 201 224
249 277 307 320]
[0134] The above array delimits (as index of Fourier spectral
lines) the frequency sub-bands of index b=0 to B-1. For example the
first sub-band (b=0) goes from the coefficient k.sub.b=0 to
k.sub.b+1-1=0; it is therefore reduced to a single coefficient
which represents 25 Hz. Likewise, the last sub-band (k=34) goes
from the coefficient k.sub.b=307 to k.sub.b+1-.sup.1=319, it
comprises 12 coefficients (300 Hz). The frequency spectral line of
index k=320 which corresponds to the Nyquist frequency is not taken
into account here.
[0135] For each frame, the ILD of the sub-band b=0, . . . , B-1 is
calculated according to equations (5) and (6) repeated here:
ILD [ b ] = 10 log 1 0 { .sigma. L 2 [ b ] .sigma. R 2 [ b ] } ( 11
) ##EQU00008##
where .sigma..sub.L.sup.2[b] and .sigma..sub.R.sup.2[b] represent
respectively the energy of the left channel (L.sub.buf[k]) and of
the right channel (R.sub.buf[k]):
{ .sigma. L 2 [ b ] = k = k b k b + 1 - 1 L [ k ] L * [ k ] .sigma.
R 2 [ b ] = k = k b k b + 1 - 1 R [ k ] R * [ k ] ( 12 )
##EQU00009##
[0136] According to a particular embodiment, the parameters ITD and
ICC are extracted in the time domain (block 320). In variants of
the invention these parameters could be extracted in the frequency
domain (block 314), this not being represented in FIG. 3 so as not
to overburden the figure. An exemplary embodiment of the estimation
of the ITD in the frequency domain is given in the standard UIT-T
G.722 Annex D on the basis of the smoothed product L[k]R*[k].
[0137] In one embodiment the parameters ITD and ICC are estimated
in the following manner. The ITD is sought by intercorrelation
according to equation (3) repeated here:
ITD=max-d.ltoreq..tau..ltoreq.d.SIGMA..sub.n=0.sup.N-.tau.-1L(n+.tau.)R(-
n) (13)
[0138] with for example d=630 .mu.s.times.F.sub.s, i.e. 10 samples
at 16 kHz. This value of 630 .mu.s is obtained for the binaural
case, on the basis of Woodworth's law defined hereinafter, with a
spherical approximation of the head (with a mean radius .alpha.=8.5
cm) and an azimuth .theta.=.pi./2.
[0139] The ITD obtained according to equation (3) is thereafter
smoothed to attenuate its temporal variations. The benefit of the
smoothing is to attenuate the fluctuations of the instantaneous ITD
which may degrade the quality of the spatial synthesis at the
decoder. The smoothing scheme adopted lies outside the scope of the
invention and it is not detailed here.
[0140] During the calculation of the ITD, the ICC is also
calculated according to equation (4) defined hereinabove.
[0141] The spatial parameters or cues ILD and ITD are coded
according to a scheme forming the subject of the invention and
described with reference to FIGS. 4a to 4c which detail the blocks
315 and 317 of FIG. 3 according to various embodiments of the
invention.
[0142] These blocks 315 and 317 implement schemes based on models
of respective representations of the cues ITD and ILD.
[0143] Certain parameters of the respective models obtained on
output from the blocks 315 and 317 are thereafter coded at 316 and
318 for example according to a scalar quantization scheme.
[0144] All the spatialization cues thus coded are multiplexed by
the multiplexer 322 before being transmitted.
[0145] Certain significant notions about sound perception are
recalled in FIGS. 5a and 5b. In FIG. 5a is illustrated a median
plane M, a frontal plane F and a horizontal plane H, with respect
to the head of a listener. Sound perception allows 3D location of a
sound source, this location is typically identified by spherical
coordinates (r, .theta., .phi.) according to FIG. 5b; in the case
of a stereo signal, perception occurs on a horizontal plane and in
this case polar coordinates (r, .theta.) suffice to locate the
source in 2D. It is also recalled that a stereo signal allows
reproduction only on a line between 2 loudspeakers on the
horizontal plane, whilst a binaural signal normally allows
perception in 3D.
[0146] In one embodiment it is considered that the signal comprises
a sound source situated in the horizontal plane.
[0147] In the case of a binaural signal, it may be useful to define
the position of a virtual source associated with the multichannel
signal to be coded. As illustrated in FIG. 5c, if one considers
only the case of a sound source 510 situated in the horizontal
plane (2D) around the person represented by a head approximated by
a sphere at 540, the position of the source is specified by the
polar coordinates (r, .theta.).
[0148] The angle .theta. is defined between the frontal axis 530 of
the listener and the axis of the source 520. The two ears of the
listener are represented as 550R for the right ear and as 550L for
the left ear. The cue in respect of time shift between the two
channels of a binaural signal is associated with the interaural
time difference, that is to say the difference in time that a sound
takes to arrive at the two ears. If the source is directly in front
of the listener, the wave arrives at the same moment at both ears
and the ITD cue is zero.
[0149] The interaural time difference (ITD) can be simplified by
using a geometric approximation in the form of the following sine
law:
ITD(.theta.)=.alpha. sin(.theta.)/c (14)
where .theta. is the azimuth in the horizontal plane, .alpha. is
the radius of a spherical approximation of the head and c the speed
of sound (in ms.sup.-1) which can be defined as c=343 ms.sup.-1.
This law is independent of frequency, and it is known to give good
results in terms of spatial location.
[0150] A virtual sound source can therefore be located with an
angle .theta. and the ITD cue can be deduced through the following
formula:
ITD(.theta.)=ITD.sub.max sin(.theta.) (15)
where
ITD.sub.max=.alpha./c (16)
The value given to ITD.sub.max may for example correspond to 630
.mu.s, which is the limit of perceptual separation between two
pulses. For larger values of ITD the subject will hear two
different sounds and will not be able to interpret the sounds as a
single sound source.
[0151] In variants of the invention the sine law could be replaced
with Woodworth's ITD model defined in the work by R. S Woodworth,
Experimental Psychology (Holt, N.Y.), 1938, pp. 520-523, by the
following equation:
ITD(.theta.)=.alpha.(sin(.theta.)+.theta.)/c (17)
which is valid for a far field (typically a source at a distance of
at least 10. .alpha.). Employing the principle of normalization by
a maximum value ITD.sub.max as in equation (15), the ITD model
according to Woodworth's law can be written in the form:
ITD ( .theta. ) = ITD max ( sin ( .theta. ) + .theta. ) 1 + .pi. /
2 where ( 18 ) ITD max = a ( 1 + .pi. / 2 ) / c ( 19 )
##EQU00010##
[0152] In variants, it would be possible to define a multiplicative
factor which does not represent the maximum value of the ITD but a
proportional value for example the factor .alpha./c. The invention
also applies in this case. For example, to simplify the expression
for Woodworth's law it is possible to write:
ITD(.theta.)=ITD.sub.max(sin(.theta.)+.theta.) (20)
where
ITD.sub.max=.alpha./c (21)
In this case the value of ITD.sub.max does not represent the
maximum value of the ITD. Hereinafter, this "disparity of notation"
will be used.
[0153] Thus, with reference to FIG. 4a, the block 315 which
receives an interchannel time shift (ITD) cue through the
extraction module 320 comprises a module 410 for obtaining a
representation model of the interchannel time shift cue.
[0154] This model is for example the model such as defined
hereinabove in equation (15) with a value ITD.sub.max=630 .mu.s
predefined in the model or the model of equation (20).
[0155] In variants, the value ITD.sub.max could be rendered
flexible by coding either this value directly, or by coding the
difference between this value and a predetermined value. This
approach makes it possible in fact to extend the application of the
ITD model to more general cases, but its drawback is to require
additional bitrate. To indicate that the explicit coding of the
value ITD.sub.max is optional, the block 412 appears dashed in FIG.
4a.
[0156] A module 411 for determining the angle .theta. such as
defined hereinabove is implemented to obtain the angle defined by
the sound source. More precisely this module searches for the
azimuth parameter .theta. which makes it possible to approach as
close as possible to the ITD extracted. When the law is known as in
equation (15), this angle can be obtained in an analytical
manner:
.theta.=.alpha. sin(ITD/ITD.sub.max) (22)
[0157] In variants, the .alpha. sin function could be
approximated.
[0158] An equivalent approach for determining the azimuth can be
implemented in the block 411. According to this approach, the
determination of the angle .theta. for the sine law calls upon a
search with the aid of the ITD model, for the closest value as a
function of the possible values of azimuth:
.theta.=argmin.sub..theta. T(ITD-ITD.sub.max sin(.theta.)).sup.2
(23)
[0159] This search can be performed by pre-storing the various
candidate values of ITD.sub.maxsin(.theta.) arising from the ITD
model in a table M.sub.ITD for a search interval which may be
T=[-.pi./2, .pi./2] assuming that the ITD is symmetric when the
source is in front of or behind the subject. In this case, the
values of .theta. are discretized, for example with a step size of
1.degree. over the search interval.
[0160] In the case of Woodworth's law, it is also possible to
follow the same approach as hereinabove for the sine law. The
analytical expression for the inverse function of
sin(.theta.)+.theta. not being trivial, it will be possible to
prefer the search:
.theta.=argmin.sub..theta.
T(ITD-ITD.sub.max(sin(.theta.)+.theta.)).sup.2 (24)
[0161] The angle parameter .theta. determined in the block 411 is
thereafter coded according to a conventional coding scheme for
example by scalar quantization on 4 bits by the block 316. This
block carries out a search for the quantization index
i=argmin.sub.j=0, . . . ,15(.theta.-Q.sub..theta.[j]).sup.2
(25)
[0162] where the table is given for the case of a uniform scalar
quantization on 4 bits
Q .theta. = { - .pi. , - 7 .pi. 8 , , 0 , .pi. 8 , , 7 .pi. 8 } (
26 ) ##EQU00011##
[0163] In variants, the number of bits allocated to the coding of
the azimuth could be different, and the quantization levels could
be non-uniform to take account of the perceptual limits of location
of a sound source according to the azimuth.
[0164] It is the coding of this parameter which makes it possible
to code the time shift cue ITD, optionally with the coding of
ITD.sub.max (block 412) as additional cue if the value
predetermined by the ITD model must be adapted. The spatialization
cue will therefore be retrieved on decoding by decoding the angle
parameter, optionally by decoding ITD.sub.max, and by applying the
same representation model of the ITD. The bitrate necessary for
coding this angle parameter is low (for example 4 bits per frame)
when no correction of the value ITD.sub.max predefined in the model
is coded. Thus, the coding of this spatialization cue (ITD)
consumes little bitrate.
[0165] At very low bitrate, the coding of a single angle .theta.
can be implemented to code the spatialization cue in respect of a
binaural signal.
[0166] In a variant embodiment, it will be possible to estimate an
ITD per frequency band, for example by taking a slicing into B
sub-bands, defined previously. In this case, an angle .theta. per
frequency band is coded and transmitted to the decoder, which for
the example of B sub-bands gives B angles to be transmitted.
[0167] In another variant, it will be possible to ignore the
estimation of the ITD for certain high frequency bands for which
the phase differences are not perceptible. Likewise, it will be
possible to omit the estimation of the ITD for very low
frequencies. For example, the ITD will not be able to be estimated
for bands above 1 kHz, and for a sub-band slicing as defined
previously it will be possible to retain the bands b=0 to 11 in the
embodiment using the 1/3 octave and 1 to 16 in the variants using
the ERB scale (the first band b=0 being omitted in the latter case
since it entails frequencies below 25 Hz). In variants of the
invention, a sub-band slicing with a different resolution from 25
Hz could be used; it will thus be possible to group together
certain sub-bands since the 1/3 octave slicing or the ERB scale may
be too fine for the coding of the ITD. This avoids coding too many
angles per frame. For each frequency band, the ITD is thereafter
converted into an angle as in the case of a single angle described
hereinabove with a bit allocation which can be either fixed or
variable as a function of the significance of the sub-band. In all
these variants where several angles are determined and coded, a
vector quantization could be implemented in the block 316.
[0168] FIG. 4b represents a variant embodiment of the invention
which can replace the mode described in FIG. 4a. The principle of
this variant is to combine in particular the blocks 411 and 316
into a block 432.
[0169] In this variant embodiment, one considers the definition of
several "competing" models for coding the ITD, knowing that the
invention also applies when a single ITD model is defined.
[0170] Thus, the model such as defined for the interchannel time
shift (ITD) cue might not be fixed and be parametrizable. Each
model defines a set of values of ITD as a function of an angle
parameter: the sine law and Woodworth's law constitute two examples
of models. In this variant, for coding, a model index and an angle
index (also called angle parameter) to be coded are determined in
the block 432 on the basis of an ITD models table obtained at 430
according to the following equation:
( m opt , t opt ) = argmin m = 0 , , N M - 1 t = 0 , , N .theta. (
m ) - 1 ( ITD - M ITD ( m , t ) ) 2 ( 27 ) ##EQU00012##
where N.sub.M is the number of models in the ITD models table,
N.sub..theta.(m) is the number of azimuth angles considered for the
m-th model and M.sub.ITD(m, t) corresponds to a precise value of
the cue ITD.
[0171] An exemplary model M.sub.ITD(m, t) is given hereinbelow in
the case of a model of index m=0 according to a Woodworth law as in
equation 20 with ITD.sub.max=0.2551 ms:
M.sub.ITD(m=1,t=0 . . . 7)=[-0.5362-0.3807-0.1978 0 0.1978 0.3807
0.5362 0.6558]
where each value is in ms. The angle index t corresponds in fact to
an angle .theta. covering the interval
] - .pi. 2 , .pi. 2 ] ##EQU00013##
with a step size of
.pi. 8 . ##EQU00014##
[0172] This table can also be referred to samples for example in
the case of a sampling at 16 kHz, one obtains in an equivalent
manner:
M.sub.ITD(m=1, t=0 . . . 7)=[-8.5795-6.0919-3.1648 0 3.1648 6.0919
8.5795 10.4930]
[0173] In this case, N.sub..theta.(m)=8 and N.sub.M=1. It is
therefore possible to code the cue ITD on 3 bits with this single
model.
[0174] It will be noted that for a given model index m, the model
M.sub.ITD (m, t) is implicitly dependent on the azimuth angle,
insofar as the index t in fact represents a quantization index for
the angle .theta.. Thus, the model M.sub.ITD t) is an efficient
means of combining the relation between ITD and .theta., and the
quantization of .theta. on N.sub..theta.(m) levels, and of
potentially using several models (at least one), indexed by
m.sub.opt when more than one model is used.
[0175] In one embodiment the case of two different models is for
example considered: [0176] m=0: A binaural model previously defined
with Woodworth's law with ITD(.theta.)=ITD.sub.max
(sin(.theta.)+.theta.) and ITD.sub.max=10 (samples at 16 kHz) m=1:
A model according to a sine law as in equation (15) but for a mic
A-B (2 omnidirectional microphones separated by a distance
.alpha.). The sine law applies here also, only the parameter
.alpha. depends on the distance between the microphones:
[0176] ITD(.theta.)=ITD.sub.max sin(.theta.) and ITD.sub.max=30
(samples at 16 kHz)
It will be noted that the size N.sub..theta.(m) may be identical
for all the models, but in the general case it is possible for
different sizes to be used. For example it will be possible to
define N.sub..theta.(m)=16 and N.sub.M=2. It is therefore possible
to code the cue ITD on 4+1=5 bits. An index of the selected law
m.sub.opt is then coded on .left brkt-top.log.sub.2N.sub.M.right
brkt-bot. bits and transmitted to the decoder in addition to the
azimuth angle t.sub.opt coded on .left
brkt-top.log.sub.2N.sub..theta..right brkt-bot. bits. In the
example taken hereinabove, it will be possible to code m.sub.opt on
1 bit, and t.sub.opt on 4 bits. In a variant, it will be possible
to replace the model m=0 by an ITD table as a function of the
azimuth arising from real measurements of HRTFs, without parametric
law, but with ITD values estimated on the real data; in this case,
the size N.sub..theta.(m) will be able to depend on the angular
resolution used to measure HRTFs (assuming that no angular
interpolation has been applied). As in FIG. 4a, the coding of a cue
in respect of correction of the value ITD.sub.max is optional, thus
the block 312 is indicated dashed. When the bit budget allocated to
the coding of ITD.sub.max is zero, the value of ITD.sub.max
predefined in the representation model of the ITD will therefore be
taken.
[0177] In a variant of the invention the representation model of
the ITD could be generalized so as to reduce solely to the
horizontal plane but also include the elevation. In this case, two
angles are determined, the azimuth angle .theta. and the elevation
angle .phi..
[0178] The search for the two angles can be made according to the
following equation:
( m opt , t opt , p opt ) = argmin m = 0 , , N M - 1 t = 0 , , N
.theta. ( m ) - 1 p = 0 , , N .PHI. ( m ) - 1 ( ITD - M ITD ( m , t
, p ) ) 2 ( 28 ) ##EQU00015##
with N.sub..phi.(m) the number of elevation angles considered for
the m-th model and p.sub.opt representing the elevation angle to be
coded.
[0179] In the invention, one also seeks to reduce the coding
bitrate of spatialization cues other than the ITD, such as the
spatialization interchannel intensity difference (ILD) cue. It will
be noted that the block 316 of FIG. 4b will be able to code and
multiplex in various ways with a fixed- or variable-bitrate coding
of the cues m.sub.opt, t.sub.opt, p.sub.opt thus ITD.sub.max as
when the latter must be transmitted.
[0180] Thus, in the same way as for the ITD it is possible to
resort to a parametrization of the ILD. In the binaural case, in
accordance with the thesis of Jerome Daniel, entitled
"Representation de champs acoustiques, application a la
transmission et a la reproduction de scenes sonores complexes dans
un contexte multimedia" [Representation of acoustic fields,
application to the transmission and reproduction of complex sound
scenes in a multimedia context], University of Paris 6, Jul. 2011,
the ILD can also be approximated according to the following
law:
ILD ( r , .theta. ) = 80 .pi. fr sin ( .theta. ) c ln ( 1 0 ) ( 29
) ##EQU00016##
[0181] where f is the frequency, r the distance from the sound
source and c the speed of sound.
[0182] By defining a relative ILD, ILD.sub.max, it is possible
under certain conditions to reduce this approximation to the
equation:
ILD.sub.glob(.theta.)=ILD.sub.max sin(.theta.) (30)
[0183] The above law is only an approximation corresponding to the
global level of the HRTFs at a given azimuth; it does not make it
possible to completely characterize the spectral coloration given
by the HRTFs but it characterizes only their global level. The
reference ILD can be defined --at a later time, when defining the
ILD model, by taking a base of normalized signals or a base of HRTF
filters--by taking the maximum of the total ILD of a binaural
signal. In the invention it is considered that this sine law
applies not only to the total (or global) ILD but also to the
sub-band based ILD; in this case, the parameter ILD.sub.max depends
on the index of the sub-band and the model becomes:
ILD[b](.theta.)=ILD.sub.max[b]sin(.theta.) (31)
[0184] Experimentally, it may be verified that if the energy
(illustrated with reference to FIG. 6a for several elevation values
.phi.) of the HRTF filters is calculated, it is apparent that the
approximation of the global ILD (in the sense of difference in
global level between channels) follows a sine law for the
elevations represented .phi.=0.degree., 15.degree. and 30.degree.,
as a function of azimuth .theta..
[0185] It will be noted that even if the symmetry of the frontal
half-plane (azimuth lying in [0, 180] degrees) and the half-plane
at the rear of the head (azimuth lying in [180, 360] degrees) is in
general not totally valid, this sine law is used in the invention
to code and decode the ILD.
[0186] Just as for the case of the ITD where a value ITD.sub.max
has been defined, it is therefore possible either to transmit the
parameter ILD.sub.max, or to use a predetermined and stored value
ILD.sub.max, so as to derive therefrom a value ILD.sub.glob
(.theta.) according to equation (30) and thus apply a global ILD,
valid over the whole spectrum of the signal to obtain a rudimentary
(global) location.
[0187] Another exemplary model relies on the configuration of ORTF
stereo microphones which is illustrated in FIG. 6b.
[0188] In this example, the sub-band based ILD model could be
defined in relation to a configuration of ORTF microphones as
follows:
ILD(.theta.)=L(.theta.)-R(.theta.)=.alpha.(cos(.theta.-.theta..sub.0)-co-
s(.theta.+.theta..sub.0) (32)
with
L(.theta.)=.alpha.(1+cos(.theta.-.theta.0)) (33)
R(.theta.)=.alpha.(1+cos(.theta.+.theta..sub.0)) (34)
where .theta..sub.0 (in radians) corresponds to 55.degree..
[0189] This model can also be written in the form:
ILD(.theta.)=L(.theta.)-R(.theta.)=.alpha.(cos(.theta.)cos(.theta..sub.0-
)+sin(.theta.)sin(.theta..sub.0)) (35)
Here again it is possible to define a value ILD.sub.max which
corresponds to:
ILD.sub.max=.alpha. (36)
Here again, it is assumed that the model defined in equation 35
applies not only to the case of a total (or global) ILD but also to
the sub-band based ILD; in this case the parameter ILD.sub.max (or
a proportional version) will be dependent on the sub-band in the
form ILD[b].sub.max.
[0190] Thus, with reference to FIG. 4a, in the same way as for the
cue ITD, the block 317 which receives an interchannel intensity
difference (ILD) cue through the extraction module 314 comprises a
module 420 for obtaining a representation model of the interchannel
intensity difference (ILD) cue.
[0191] This model is for example the model such as defined
hereinabove in equation (30) or with other models described in this
document.
The angle parameter .theta. already defined at 411 can be reused at
the decoder to retrieve the global ILD or the sub-band based ILD
such as defined by equation (30), (31) or (35); this in fact makes
it possible to "pool" the coding of the ITD and of the ILD. In the
case where the value ILD.sub.max is not fixed, the latter is
determined at 423 and coded.
[0192] In a particular embodiment, a module 421 for estimating an
interchannel intensity difference cue is implemented on the basis
on the one hand of the angle parameter obtained by the block 411 in
order to code the time shift cue (ITD) and on the other hand of the
representation model of equation (30), (31) or (35). In an optional
manner, the module 422 calculates a residual of the cue ILD, that
is to say the difference between the cue in respect of real
interchannel intensity difference (ILD) extracted at 314 and the
interchannel intensity difference (ILD) cue estimated at 421 on the
basis of the ILD model.
[0193] This residual can be coded at 318 for example by a
conventional scalar quantization scheme. However, in
contradistinction to the coding of a direct ILD, the quantization
table may be for example limited to a dynamic range of +/-12 dB
with a step size of 3 dB.
[0194] This ILD residual makes it possible to improve the quality
of decoding of the cue ILD in the case where the ILD model is too
specific and applies only to the signal to be coded in the current
frame; it is recalled that a classification may optionally be used
at the coder to avoid such cases, however in the general case it
may be useful to code an ILD residual.
[0195] Thus, the coding of these parameters as well as that of
angle of the ITD makes it possible to retrieve at the decoder the
interchannel intensity difference (ILD) cue of the binaural audio
signal with a good quality.
[0196] In the same way as for the ITD, the spatialization cue
(global or sub-band based) will therefore be retrieved on decoding
by applying the same representation model and by decoding if
relevant the residual parameter and reference ILD parameter. The
bitrate necessary for coding these parameters is lower than if the
cue ILD itself were coded, in particular when the ILD residual does
not have to be transmitted and when use is made of the parameter or
parameters ILD.sub.max predefined in the ILD model or models. Thus,
the coding of this spatialization cue (ILD) consumes little
bitrate.
[0197] This ILD model using only a global ILD value is however very
simplistic since in general the ILD is defined on several
sub-bands.
[0198] In the coder described previously, B sub-bands according to
a 1/3 octave slicing or according to the ERB scale were defined. To
make it possible to represent more than one parameter of total (or
global) ILD the representation model of the ILD is therefore
extended to several sub-bands. This extension applies to the
invention described in FIG. 4a, however the associated description
is given hereinafter in the context of FIG. 4b to avoid too much
redundancy. The model is dependent on the angle .theta. and
optionally on the elevation; this model may be the same in all the
sub-bands, or vary according to the sub-bands.
[0199] We consider the variant embodiment described in FIG. 4b for
the coding of the ILD. Just as for the ITD, in this variant we
define representation models of the ILD. The model such as defined
for the interchannel intensity difference (ILD) cue is not fixed
but is parametrizable. The model is defined by a value ILD.sub.max
and an angle parameter. In the general case, on the basis of an ILD
models table obtained at 440, we determine a model index m.sub.opt
and an angle index to be coded at 442 according to the following
equation:
( m opt , t opt ) = argmin m = 0 , , N M - 1 t = 0 , , N .theta. (
m ) - 1 dist ( ILD , M ILD ( m , t ) ) ( 37 ) ##EQU00017##
[0200] where N.sub.M is the number of models in the ILD models
table, N.sub..theta.(m) is the number of azimuth angles considered
for the m-th model, M.sub.ILD (m, t) corresponds to a precise value
of the cue ILD and dist(.,.) is a criterion of distance between ILD
vectors. However, in a variant embodiment, this search could be
simplified by using the angle cue already obtained in the block 432
for the ITD model. It will be noted that the values t=0, . . . ,
N.sub..theta.(m)-1 for the ILD model do not necessarily correspond
to the same set of values as for the ITD model, however it is
advantageous to harmonize these sets so as to have coherence
between representation models for the ILD and the ITD.
[0201] The following may for example be taken as possible distance
criteria:
dist(X,Y)=|.SIGMA..sub.b=0.sup.B-1X[b]-.SIGMA..sub.b=0.sup.B-1Y[b]|.sup.-
q (38)
where q=1 or 2.
[0202] An exemplary ILD model is illustrated in FIGS. 6c to 6g for
several frequency bands. We do not give here the corresponding
values (in dB) in the form of arrays so as not to overburden the
text, approximate values could be derived from the graphs of FIGS.
6c to 6g. This figure considers the case of a 1/3 octave slicing
already defined previously. Thus each figure represents the ILD for
the frequency band defined by the octave-third number defined in
the array 1 hereinabove with a band-dependent central frequency fc.
Each point marked with a circle in each sub-figure corresponds to a
value M.sub.ILD(m, t); in addition to defining the ILD table
associated with the model we have also shown the sine law scaled by
a predefined parameter ILD.sub.max dependent on the sub-band.
[0203] In a variant of the invention the representation model of
the ILD could be generalized so as not to reduce solely to the
horizontal plane but also to include the elevation. In this case,
the search for two angles becomes:
( m opt , t opt , p opt ) = argmin m = 0 , , N M - 1 t = 0 , , N
.theta. ( m ) - 1 p = 0 , , N .PHI. ( m ) - 1 dist ( ILD , M ITD (
m , t , p ) ) ( 39 ) ##EQU00018##
with N.sub..phi.(m) the number of elevation angles considered for
the m-th model and p.sub.opt representing the elevation angle to be
coded.
[0204] In a variant, an exemplary model M.sub.ILD (m, t, p) can be
obtained on the basis of a suite of HRTFs in the following manner.
Given the HRTF filters for .theta. and .phi., it is possible to:
[0205] calculate the ILDs per sub-band between left and right
channels per sub-band [0206] optionally normalize the ILDs [0207]
store the ILDs and determine the value of ILD.sub.max in each
sub-band so as to adjust an expansion factor for the ILDs The
multidimensional table M.sub.ILD (m, t, p) can be seen as a
directivity model referred to the domain of the ILD.
[0208] An index of the selected law m.sub.opt is then coded and
transmitted to the decoder at 318.
In the same way as for FIG. 4a, an ILD residual could be calculated
(blocks 421 and 422) and coded.
[0209] Hitherto separate models have been considered for the ITD
and the ILD, even if it was noted that the determination of the
angle may be "pooled". For example, the azimuth may be determined
by using the ITD model and this same angle is used directly for the
ILD model. Another variant embodiment calling upon a (joint)
"integrated model" is now considered. This variant is described in
FIG. 4c.
[0210] In this variant, rather than having separate models for the
ITD and the ILD (M.sub.ITD (m, t, p) and M.sub.ILD (m, t, p)) it
will be possible to define a joint model in the block 450:
M.sub.ITD,ILD (m, t, p) whose inputs comprise candidate values of
ITD and of ILD; thus, for various discrete values representing
.theta. and .phi. "vectors" (ITD, ILD) are defined. In this case,
the distance measurement used for the search must combine the
distance on the ITD and the distance on the ILD, however it is
still possible to perform a separate search.
[0211] Thus, an index of the selected law m.sub.opt, of the azimuth
angle t.sub.opt and of the elevation angle p.sub.opt that are
determined at 453, are coded at 331 and transmitted to the decoder.
Just as for FIGS. 4a and 4b, the parameters ITD.sub.max,
ILD.sub.max and the ILD residual can be determined and coded.
[0212] A variant of the coder illustrated in FIG. 3 implementing
the joint model of FIG. 4c is illustrated in FIG. 8. It will be
noted that in this coder variant the parameters ITD and ICC are
estimated in the block 314. Moreover, here we consider the general
case where IPD parameters are also extracted and coded in the block
332. The blocks 330 and 331 correspond to the blocks indicated and
detailed in FIG. 4c.
[0213] With reference to FIG. 7 a decoder according to one
embodiment of the invention is now described.
[0214] This decoder comprises a demultiplexer 701 in which the
coded mono signal is extracted so as to be decoded at 702 by a mono
EVS decoder (according to the specifications 3GPP TS 26.442 or TS
26.443) in this example. The part of the binary train corresponding
to the mono EVS coder is decoded according to the bitrate used at
the coder. It is assumed here that there is no loss of frames nor
any binary errors in the binary train to simplify the description,
however known techniques for correcting loss of frames can quite
obviously be implemented in the decoder.
[0215] The decoded mono signal corresponds to {circumflex over
(M)}(n) in the absence of channel errors. An analysis by short-term
discrete Fourier transform with the same windowing as at the coder
is carried out on {circumflex over (M)}(n) (blocks 703 and 704) to
obtain the spectrum {circumflex over (M)}[k]. It is considered here
that a decorrelation in the frequency domain (block 720) is also
applied. This decorrelation could also have been applied in the
time domain.
[0216] The details of implementation of the block 708 for the
synthesis of the stereo signal are not presented here since they
lie outside the scope of the invention, but the conventional
synthesis techniques known from the prior art could be used.
In the synthesis block 708, it is for example possible to
reconstruct a signal with two channels with the following
processing on the mono signal decoded and transformed into
frequencies:
{circumflex over (L)}[k]=c.sub.1{circumflex over (M)}[k] (40)
{circumflex over (R)}[k]=c.sub.2{circumflex over
(M)}[k]e.sup.-j2.pi.kiTD/NFFT (41)
where c=10.sup.ILD[b]/10 (with b the index of the sub-band
containing the spectral line of index k),
c 1 = 2 c 1 + c , and ( 42 ) c 2 = 2 1 + c , ( 43 )
##EQU00019##
ITD is the ITD decoded for the spectral line k (if a single ITD is
coded, this value is identical for the various spectral lines of
index k) and NFFT is the length of the FFT and of the inverse FFT
(blocks 704, 709, 712). It is also possible to take into account
the parameter ICC decoded at 718 to recreate a non-localized sound
ambience (background noise) to improve the quality.
[0217] The spectra {circumflex over (L)}[k] and {circumflex over
(R)}[k] are thus calculated and thereafter converted into the time
domain by inverse FFT, windowing, addition and overlap (blocks 709
to 714) to obtain the synthesized channels {circumflex over (L)}(n)
and {circumflex over (R)}(n).
[0218] The parameters which have been coded to obtain the
spatialization cues are decoded at 705, 715 and 718.
[0219] At 718, it is the cues ICC.sup.q[b] which are decoded if,
however, they have been coded.
[0220] At 705, it is the angle parameter .theta. which is decoded,
optionally with a value ITD.sub.max. On the basis of this
parameter, the module 706 for obtaining a representation model of
an interchannel time shift cue is implemented to obtain this model.
Just as for the coder, this model can be defined by equation (15)
defined hereinabove. Thus, on the basis of this model and of the
decoded angle parameter, it is possible for the module 707 to
determine the interchannel time shift (ITD) cue in respect of the
multichannel signal.
[0221] If at the decoder an angle per frequency or per frequency
band is coded, then these various angles per frequency or frequency
bands are decoded to define the cues ITD per frequency or frequency
bands.
[0222] In the same way, in the case where parameters making it
possible to code the interchannel intensity difference (ILD) cue
are coded, they are decoded by the module for decoding these
parameters at 715, at the decoder.
[0223] Thus, the residual parameter (Resid. ILD) and reference ILD
parameter (ILD.sub.max) are decoded at 715.
[0224] On the basis of these parameters, the module 716 for
obtaining a representation model of an interchannel intensity
difference cue is implemented to obtain this model. Just as for the
coder, this model can be defined by equation (30) defined
hereinabove.
[0225] Thus, on the basis of this model, of the ILD residual
parameters (that is to say the difference between the cue in
respect of real interchannel intensity difference (ILD) and the
interchannel intensity difference (ILD) cue estimated with the
model), of the reference ILD parameter (ILD.sub.max) and of the
angle parameter decoded at 705 for the cue ITD, it is possible for
the module 717 to determine the interchannel intensity difference
(ILD) cue of the multichannel signal.
[0226] If at the coder the ILD coding parameters were itemized by
frequency band, then these various frequency band based parameters
are decoded to define the cues ILD per frequency or frequency
bands.
[0227] It will be noted that the decoder of FIG. 7 is relevant to
the coder of FIG. 4a. It will be understood that if the coding
according to the invention is done according to FIG. 4b or 4c, the
decoder will be modified accordingly to decode, in particular,
indices of models and of angles in the form m.sub.opt, t.sub.opt,
p.sub.opt and to reconstruct the values of ITD and of ILD as a
function of the model used and indices associated with
reconstruction values.
[0228] In a variant of the invention the decoder of FIG. 7 is thus
modified as illustrated in FIG. 9. In this variant, the decoded ILD
and ITD parameters are not reconstructed directly. The stereo
synthesis (block 708) is replaced with a binaural synthesis (block
920). Thus the decoding of the cues ILD and ITD reduces to a
decoding (block 910) of the angular coordinates. By using a
predefined basis of HRTFs (block 930) it is therefore possible to
decode a binaural signal rather than a stereo signal. In variants,
it will be possible to apply the HRTF filters in the time
domain.
[0229] The coder presented with reference to FIG. 3 and the decoder
presented with reference to FIG. 7 have been described in the case
of particular application of stereo coding and decoding. The
invention has been described on the basis of a decomposition of the
stereo channels by discrete Fourier transform. The invention also
applies to other complex representations, such as for example the
MCLT (Modulated Complex Lapped Transform) decomposition combining a
modified discrete cosine transform (MDCT) and modified discrete
sine transform (MDST), as well as in the case of banks of filters
of Pseudo-Quadrature Mirror Filter (PQMF) type. Thus the term
"frequency spectral line" used in the detailed description can be
extended to the notion of "sub-band" or of "frequency band",
without changing the nature of the invention.
The coders and decoders such as described with reference to FIGS. 3
and 7 can be integrated into multimedia equipment of lounge
decoder, "set top box" or audio or video content reader type. They
can also be integrated into communication equipment of mobile
telephone or communication gateway type.
[0230] FIG. 10 represents an exemplary embodiment of such an item
of equipment in which a coder such as described with reference to
FIGS. 3, 8 and 4a to 4c or a decoder such as described with
reference to FIG. 7 or 9, according to the invention is integrated.
This device comprises a processor PROC cooperating with a memory
block BM comprising a storage and/or work memory MEM.
[0231] In the case of a coder, the memory block can advantageously
comprise a computer program comprising code instructions for the
implementation of the steps of the coding method in the sense of
the invention, when these instructions are executed by the
processor PROC, and in particular the steps of extracting a
plurality of spatialization cues in respect of the multichannel
signal, of obtaining at least one representation model of the
spatialization cues extracted, of determining at least one angle
parameter of a model obtained and of coding the at least one angle
parameter determined so as to code the spatialization cues
extracted during the coding of spatialization cues.
[0232] In the case of a decoder, the memory block can
advantageously comprise a computer program comprising code
instructions for the implementation of the steps of the decoding
method in the sense of the invention, when these instructions are
executed by the processor PROC, and in particular the steps of
receiving and decoding at least one coded angle parameter, of
obtaining at least one representation model of spatialization cues
and of determining a plurality of spatialization cues in respect of
the multichannel signal on the basis of the at least one model
obtained and of the at least one decoded angle parameter.
[0233] The memory MEM can store the representation model or models
of various spatialization cues which are used in the coding and
decoding methods according to the invention.
[0234] Typically, the descriptions of FIGS. 3, 4 on the one hand
and 7 on the other hand repeat the steps of an algorithm of such a
computer program respectively for the coder and for the decoder.
The computer program can also be stored on a memory medium readable
by a reader of the device or item of equipment or downloadable into
the memory space of the latter.
[0235] Such an item of equipment in the guise of coder comprises an
input module able to receive a multichannel signal for example a
binaural signal comprising the channels R and L for right and left,
either through a communication network, or by reading a content
stored on a storage medium. This multimedia equipment item can also
comprise means for capturing such a binaural signal.
[0236] The device in the guise of coder comprises an output module
able to transmit a mono signal M arising from a channels reduction
processing and at the minimum, an angle parameter .theta. making it
possible to apply a representation model of a spatialization cue so
as to retrieve this spatial cue. If relevant, other parameters such
as the ILD residual, ILD or reference ITD (ILDmax or ITDmax)
parameters are also transmitted via the output module.
[0237] Such an item of equipment in the guise of decoder comprises
an input module able to receive a mono signal M arising from a
channels reduction processing and at the minimum an angle parameter
.theta. making it possible to apply a representation model of the
spatialization cue so as to retrieve this spatial cue. If relevant,
to retrieve the spatialization cue, other parameters such as the
ILD residual, ILD or reference ITD (ILDmax or ITDmax) parameters
are also received via the input module E.
[0238] The device in the guise of decoder comprises an output
module able to transmit a multichannel signal for example a
binaural signal comprising the channels R and L for right and
left.
[0239] Although the present disclosure has been described with
reference to one or more examples, workers skilled in the art will
recognize that changes may be made in form and detail without
departing from the scope of the disclosure and/or the appended
claims.
* * * * *